| 3c089d751ede283e21e186885eaf705c3257ccd2 |
|
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| 966a708b93795de9e22df632011af76f00b6d0a7 |
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17-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Use RtpFileSource in NetEqDecodingTest
This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.
BUG=2692
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| 4b133da5fd37a93de2f191ef340fd105e6f83672 |
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02-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Let RtpFileSource use RtpFileReader
RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.
All NetEq test tools that use RtpFileSource are updated.
BUG=2692
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| c8e98187d1751a0ed31a0e76ea80564c4e4a4c04 |
|
26-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Receiver bit-exactness test for AudioCoding Module
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.
The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| 12396aba4262a03dbdaf9fce3e6bedbfaad7e86d |
|
18-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update PacketSource and RtpFileSource
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| 9c55f0f957534144d2b8a64154f0a479249b34be |
|
09-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| 1b9df05c8521d1d807b08d7c00eb2f7e5b097fdf |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6257 "Rename neteq4 folder to neteq"
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
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| a90f6d67f72359cf63b59480fa87a13aae808c03 |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/packet_source.h
|